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If it's a Cisco switch you can search the mac address table >sh mac address table | inc xxxx where xxxx is the last 4 of the phone's mac address

Actually just spotted your 1200/1300 range in the OP I'd run a wireshark to see if the IPO is requesting the port change or what port it is sending the traffic to the watchguard over. Does sound like some weird custom NAT going on

What port/s is it changing to? SIP can use a range of 5056 - 5071 but it's generally the common ones of 5056/5060/5061 used by most vendors.

double check the signalling group for TG1 and perhaps try to recreate it. For the 408 I'd check the phones are registered OK and see if there have been any firewall changes around the time this started happening.

would need more info is CM simplex/duplex what hardware is CM are handsets static IP or DHCP

Have you setup the ANPS apple notification push service in SM? Can't remember off the top of my head but you need a endpoint policy or similar in SBC to send/proxy the APN to the IoS device. There used to be a good guide on setting up the IX workplace client on the SBC for remote workers

If it's Vodafone, whatever config they tell you they have done the router will be set up differently [mad]

It shouldn't matter if they plug a handset or computer into a PoE enabled network port. The device will negotiate the PoE with the switch and if not required the switch doesn't activate it. If you are using Avaya you can plug the computer into the 9608 handset as it has a builtin switch, you...

You need to import the godaddy Root CA to the SBCE as well. You also need to generate the CSR on the SBCE as it also creates a key file used for the import. I found this guide good for SBCE certificates, I did an SMGR ident cert for the inside interface and a Globalsign public cert for the...

most of those flat ethernet cables are stranded core which is more difficult to crimp. Try and find solid core for making DIY patch cables.

Platform name is what you want to call it The FQDN/IP field needs to be the fqdn name of the server i.e ntq----01.kmph.com You also need to ensure that fqdn is resolvable via DNS i.e you have an A and ptr record in DNS for it (or use the host file on your client) You can get the cert common...

The master licence is in PLDS Don't know R10 but as it's just a vmware licence so you could register an account with vmware add the licence and vmware will allow you to upgrade it to R8.0 for free. The licence can be loaded/installed via the esxi web interface Navigator >> host >> manage >>...

The handset should remember the last extension and register with that after a disconnection, it certainly did with our 9608 handsets after a shut/no shut of the PoE port or switch reset. What handsets are you using and are they H.323 or SIP?

I've just installed IPO v11 and set it to demo mode with the demo config. Installed the latest One-X communicator from PLDS and set it to H.3232 and server IP When I log in it says "please wait while one-x is launching" spinning wheel stops and then the app just closes. If I put the wrong...

Look at what is causing the service interruptions and fix that. If the phones are going into discovery that sounds like a networking issue.

Yes these steps 1. contact an avaya installation or business partner 2. ask for a quote to install a G450 3. pay them

We still use AC15 and analogue MFCR2 on our air traffic voice system. ISDN30/Megastream has a massive advantage over SIP in the call setup time and latency but support from BT/Openreach is getting worse (and slower) by the year as spares and engineers become fewer.

Ah looks like I was wrong about dual streams, looks like DMCC MR sums the streams into a single one by default. Found this for AES https://documentation.avaya.com/bundle/AdministeringAEServices_r8.1/page/Media_Content_and_Media_Tone_Announcement_for_DMCC.html and the NICE manual also says...

Nope, if it's an IP recorder it gets the audio as two streams and records it as stereo anyway. Have a look at the Media drive and the latest Calls folder and make a test call, you should see 2x WAV files and an XML while the call is in progress and after hang-up it will change to a single WAV...

Check the recorder compression settings. If you have "mixed mono" enabled it will combine the stereo stream files into a single mono one